Voice communication is evolving from circuit-switched technology, such as provided by the Public Switched Telephone Network (PSTN) or Public Land Mobile Network (PLMN), to packet-switched technology, such as provided by Voice over IP (VoIP) techniques across IP Networks. Indeed, the Internet Engineering Task Force (IETF) has developed IP-based protocols to perform various functions in VoIP communications. In particular, for example, Session Initiation Protocol (SIP), specified in IETF request for comment document RFC 2543, has been developed for establishing voice calls between two parties. In contrast, Real-time Transport Protocol (RTP), specified in IETF request for comment document RFC 1889, has been developed to format packetized voice to be carried over the Internet once the call has been established.
At the same time as voice communication is evolving, wireless networks are evolving from circuit-switched voice networks (e.g., GSM, IS-136, IS-95) to packet-switched networks (e.g., WLAN, UMTS, cdma2000) capable of supporting multimedia applications to mobile end-users over IP. General Packet Radio Service (GPRS), which is an evolution of GSM, can support packet data (e.g., web browsing, email) in a cellular environment. Further evolution of GPRS, often referred to as the Universal Mobile Telecommunication System (UMTS), is expected to support real-time multimedia over IP (e.g., VoIP, video over IP, streaming media) in a cellular environment. In addition, the Third Generation Partnership Project (3GPP) has specified the IP Multimedia Subsystem (IMS) in UMTS to accomplish the control and service functions of wireless IP multimedia. In this regard, the 3GPP has adopted SIP as the signaling protocol in IMS. At the same time, in the cdma2000 world, the 3GPP2 has been developing the IP Multimedia Subsystem (IMS), formerly referred to as the IP Multimedia Domain (MMD), to implement the control and service functions of wireless IP multimedia. The 3GPP2 has also adopted SIP into the IMS specification.
The mass deployment of IP networks supporting VoIP is expected to happen in the future. At first, VoIP is expected to penetrate into fixed (desktop) phone and wireless LAN (WLAN) segments of the communications industry, followed by penetration into the cellular (3G) segment. Thus, two kinds of heterogeneity are conceivable in the near future. In one dimension, there will be a large number of VoIP phones, as well as a large number of PSTN phones. To address this heterogeneity, provisions are made in SIP to allow a VoIP phone to call a telephone number in PSTN and vice versa.
In another dimension, footprints of wireless access networks that are enabled with VoIP (e.g., WLAN, UMTS, cdma2000) will overlap with footprints of those using traditional circuit switched technology for voice (e.g., GSM, GPRS, IS-136, IS-95). In this regard, consider a mobile node that has two interfaces, such as WLAN and GSM. Assume that the mobile node is currently in the coverage of an indoor WLAN network and engaged in a voice call over a WLAN interface using VoIP with a correspondent node who is using a traditional (non-IP) telephone connected to the PSTN. In this regard, provisions exist within SIP to enable establishment of such a call, a part of which spans IP network and a part the PSTN. When the mobile node moves out of the coverage of WLAN network, however, the call is typically dropped. Dropping such calls, as will be appreciated, is often an annoyance to end users since they have to initiate a new call to resume the voice call. However, there are currently no techniques, to best knowledge of the inventors, that allow the user in an IP access network to undergo handoff to a circuit-switched access network without interrupting the voice call.